to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
They are highly convenient for writing configuration files, especially
if you have many similar entries, or want to switch quickly between
different configurations without having to comment/uncomment large
sections of the files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.
Issue #7989, patch by DEA, slightly modified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.
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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Implement option for allow/disallow subscriptions
- Implement option for allow/disallow overlap dialling
- Set default to disable overlap dialling in sip.conf.sample for new installations
- Remove overlap dialling from subscription logic
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
